chromium/third_party/webrtc/modules/audio_processing/audio_processing_impl.h

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_

#include <stdio.h>

#include <atomic>
#include <list>
#include <memory>
#include <string>
#include <vector>

#include "absl/base/nullability.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/audio/audio_processing.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/function_view.h"
#include "api/task_queue/task_queue_base.h"
#include "modules/audio_processing/aec3/echo_canceller3.h"
#include "modules/audio_processing/agc/agc_manager_direct.h"
#include "modules/audio_processing/agc/gain_control.h"
#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/capture_levels_adjuster/capture_levels_adjuster.h"
#include "modules/audio_processing/echo_control_mobile_impl.h"
#include "modules/audio_processing/gain_control_impl.h"
#include "modules/audio_processing/gain_controller2.h"
#include "modules/audio_processing/high_pass_filter.h"
#include "modules/audio_processing/include/aec_dump.h"
#include "modules/audio_processing/include/audio_frame_proxies.h"
#include "modules/audio_processing/ns/noise_suppressor.h"
#include "modules/audio_processing/render_queue_item_verifier.h"
#include "modules/audio_processing/rms_level.h"
#include "rtc_base/gtest_prod_util.h"
#include "rtc_base/swap_queue.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"

namespace webrtc {

class ApmDataDumper;
class AudioConverter;

constexpr int RuntimeSettingQueueSize() {}

class AudioProcessingImpl : public AudioProcessing {};

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_